Hi, The following are a few RFC snippets. As shown within RFC 3725 section 10.2, 3PCC is one of the users of re-INVITE without SDP. If a device does not support re-INVITE without SDP or doesn't offer all codecs that the UA is currently willing and able to use, it hinders interoperability during 3PCC situations. For instance within RFC 3725 figure 13, the Media Server may not be unable to communicate with the Pre-Paid User if the prepaid device limits the offered codecs to be whatever was recently negotiated with the Called Party.
RFC 3261 section 14.2: "A UAS providing an offer in a 2xx (because the INVITE did not contain an offer) SHOULD construct the offer as if the UAS were making a brand new call, subject to the constraints of sending an offer that updates an existing session, as described in [13] in the case of SDP. Specifically, this means that it SHOULD include as many media formats and media types that the UA is willing to support." RFC 6337 section 5.1: "A UA should send an offer that indicates what it, and its user, are interested in using/doing at that time, without regard for what the other party in the call may have indicated previously. This is the case even when the offer is sent in response to an INVITE or re-INVITE that contains no offer. (However, in the case of re-INVITE, the constraints of [RFC3261] and [RFC3264] must be observed.)" RFC 6337 section 5.2.5: "When the new offer is sent in response to an offerless (re-)INVITE, it should be constructed according to the General Principle for Constructing Offers and Answers (Section 5.1 ): all codecs the UA is currently willing and able to use should be included, not just the ones that were negotiated by previous offer/answer exchanges. The same is true for media types -- so if UA A initially offered audio and video to UA B, and they end up with only audio, and UA B sends an offerless (re-)INVITE to UA A, A's resulting offer should most likely re-attempt video, by reusing the zeroed "m=" line used previously." RFC 6337 section 5.3: "If a UA has occasion to send another offer in the session, without any desire to change the hold status (e.g., in response to a re-INVITE without an offer, or when sending a re-INVITE to refresh the session timer), it should follow the "General Principle for Constructing Offers and Answers" (Section 5.1). If it previously initiated a "hold" by sending "a=sendonly" attribute or "a=inactive" attribute, then it should offer that again. If it had not previously initiated "hold", then it should offer "a=sendrecv" attribute, even if it had previously been forced to answer something else. Without this behavior it is possible to get "stuck on hold" in some cases, especially when a 3pcc is involved." > -----Original Message----- > From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip- > implementors-boun...@lists.cs.columbia.edu] On Behalf Of Ramesh Babu Kuppili > Sent: Monday, January 18, 2016 3:06 PM > To: sip-implementors@lists.cs.columbia.edu > Subject: [Sip-implementors] Codec negotiation when incoming re-INVITE has no > SDP > > Hello SIP experts, > > I have a question about codec negotiation when a UA receives re-INVITE with > no SDP. > > Lets say, the codec negotiation between UA and Proxy ended up with G711 and > RFC 2833 (m=audio 60146 RTP/AVP 0 101). However after the call gets > established, if the proxy sends re-INVITE with no SDP, what should be the > codec list in the "200 OK" response for the reINVITE. My proxy vendor claims > that the "200 OK" should contain the list of the codecs supported by the UA. > Not the codec list already negotiated in the dialogue. Can someone point me > in the right direction for this scenario as to what the RFC says and if > possible a section number in the RFC. > > UA Proxy > -----------------------------------------> > (INVITE SDP: codecs 0 18 101) > > <------------------------------------------ > (200 OK SDP: codecs 0 101) > > --------------------------------------------> > ACK (no SDP) > > > <------------------------------------------- > (re-INVITE no SDP) > > ----------------------------------------------> > (200 OK SDP: codecs ?????) > > - ramesh _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
